On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.g...@gmail.com> wrote:
> I am trying to get audio to work on AWS using asterisk 18.14.0 > > I have enabled the firewall to allow ALL UDP on AWS > > My SIP extension has > nat=force_rport,comedia > qualify=yes > allow=ulaw > allow=alaw > allow=gsm > canreinvite=yes > > I enable "rtp set debug on" and the console is printing info. > > The call comes into my linphone softphone - but I get no audio on my > linphone softphone. > What might I be missing to allow the audio ? > Volume is up. > > Thanks > > Jerry > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan address of the linphone - it should be sending to the NAT address and is not. What did I not set correctly ? I am not using pjsip - but the older asterisk. Thanks Jerry
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